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<td id="header-avatar-image-container" valign="top" style="padding: 0px; border-collapse: collapse; vertical-align: top; width: 32px; padding-right: 9px"><a href="https://wiki.asterisk.org/wiki/display/~gjoseph?src=email" style="color: #3b73af; text-decoration: none"><img id="header-avatar-image" class="image_fix" src="cid:avatar_56b2a47747d2721d6dcf26979b4b605a" height="32" width="32" border="0" style="border-radius: 3px; vertical-align: top" /></a></td>
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<h1 id="ConfiguringAsteriskforWebRTCClients-Overview" style="margin: 10px 0 0 0; margin-top: 0; font-size: 24px; font-weight: normal; line-height: 30px; margin: 40px 0 0 0; margin-top: 0">Overview</h1>
<p style="margin: 10px 0 0 0">This tutorial will walk you through configuring Asterisk to service WebRTC clients.</p>
<p style="margin: 10px 0 0 0">You will...</p>
<ul style="margin: 10px 0 0 0">
<li>Modify or create an Asterisk HTTPS TLS server.</li>
<li>Create a PJSIP WebSocket transport.</li>
<li>Create PJSIP Endpoint, AOR and Authentication objects that represent a WebRTC client.</li>
</ul>
</div> </td>
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<p style="margin: 10px 0 0 0; margin-top: 0"></p>
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<th class="diff-macro-title" style="background-color: transparent; text-align: left; font-weight: normal;padding: 5px;"><span class="icon macro-placeholder-icon" style="background-color: ;line-height: 20px;"><img src="https://wiki.asterisk.org/wiki/s/en_GB/5639/a252d7f5e75d7a8bf7047b4b2c92f71a56a8f048.48/_/images/icons/macrobrowser/dropdown/toc.png" style="padding-right: 5px; vertical-align: text-bottom;" /> </span>Table of Contents</th>
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<td style="background-color: #fafafa; padding: 0 0 0 5px; font-size: 12px; text-align: left;; padding: 0px; border-collapse: collapse">printable</td>
<td style="background-color: #fafafa; padding: 0 0 0 5px; font-size: 12px; text-align: left;; padding: 0px; border-collapse: collapse">false</td>
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<td style="background-color: #fafafa; padding: 0 0 0 5px; font-size: 12px; text-align: left;; padding: 0px; border-collapse: collapse">true</td>
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<p style="margin: 10px 0 0 0"></p>
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<h1 id="ConfiguringAsteriskforWebRTCClients-Prerequisites" style="margin: 10px 0 0 0; margin-top: 0; font-size: 24px; font-weight: normal; line-height: 30px; margin: 40px 0 0 0; margin-top: 0">Prerequisites</h1>
<p style="margin: 10px 0 0 0"> <span style="color: rgb(133,120,102);font-family: MuseoSans500 , sans-serif;font-size: 20.0px;">Asterisk Installation:</span> </p>
<p style="margin: 10px 0 0 0">You should have a working chan_pjsip based Asterisk installation to start with and for purposes of this tutorial, it must be version 15.5 or greater. Either install Asterisk from your distribution's packages or preferably <a class="confluence-link unresolved" href="#" style="color: #3b73af; text-decoration: none">install Asterisk from source</a>. Either way, there are a few modules over and above the standard ones that must be present for websockets and WebRTC to work:</p>
<ul style="margin: 10px 0 0 0">
<li>res_crypto</li>
<li>res_http_websocket</li>
<li>res_pjsip_transport_websocket</li>
<li>codec_opus (optional but highly recommended for high quality audio)<br /> <br /> </li>
</ul>
<div>
We recommend installing Asterisk from source because it's easy to make sure those modules are built and installed.
</div>
<h2 id="ConfiguringAsteriskforWebRTCClients-Certificates:" style="margin: 10px 0 0 0; font-size: 20px; font-weight: normal; line-height: 30px; margin: 40px 0 0 0">Certificates:</h2>
<p style="margin: 10px 0 0 0">Technically, a client can use WebRTC over an unsecured websocket to connect to Asterisk. In practice though, most browsers will require a TLS based websocket to be used. You can use self-signed certificates to set up the Asterisk TLS server but getting browsers to accept them is tricky so if you're able, we highly recommend getting trusted certificates from an organization such as <a href="https://letsencrypt.org" class="external-link" rel="nofollow" style="color: #3b73af; text-decoration: none">LetsEncrypt</a>.</p>
<p style="margin: 10px 0 0 0">If you already have certificate files (certificate, key, CA certificate), whether self-signed or trusted, you can skip the rest of this section. If you need to generate a self-signed certificate, read on.</p>
<h3 id="ConfiguringAsteriskforWebRTCClients-CreateCertificates" style="margin: 10px 0 0 0; font-size: 16px; line-height: 25px; margin: 30px 0 0 0"> <span style="color: rgb(133,120,102);">Create Certificates</span> </h3>
<p style="margin: 10px 0 0 0">Asterisk provides a utility script, <strong>ast_tls_cert</strong> in its <strong>contrib/scripts</strong> source directory. We will use it to make a self-signed certificate authority and a server certificate for Asterisk, signed by our new authority.</p>
<p style="margin: 10px 0 0 0">From the Asterisk source directory run the following commands. You'll be prompted to set a a pass phrase for the CA key, then you'll be asked for that same pass phrase a few times. Use anything you can easily remember. The pass phrase is indicated below with "********". Replace "pbx.example.com" with your PBX's hostname or ip address. Replace "My Organization" as appropriate.</p>
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<td class="diff-macro-body" style="background-color: #fff;border: 1px solid #dddddd;padding: 10px;; padding: 0px; border-collapse: collapse"> <pre style="margin: 10px 0 0 0; margin-top: 0">$ sudo mkdir /etc/asterisk/keys
$ sudo contrib/scripts/ast_tls_cert -C pbx.example.com -O "My Organization" -d /etc/asterisk/keys
No config file specified, creating '/etc/asterisk/keys/tmp.cfg'
You can use this config file to create additional certs without
re-entering the information for the fields in the certificate
Creating CA key /etc/asterisk/keys/keys/ca.key
Generating RSA private key, 4096 bit long modulus
............................................................................++
.....................++
e is 65537 (0x010001)
Enter pass phrase for /etc/asterisk/keys/ca.key:********
Verifying - Enter pass phrase for /etc/asterisk/keys/ca.key:********
Creating CA certificate /etc/asterisk/keys/ca.crt
Enter pass phrase for /etc/asterisk/keys/ca.key:********
Creating certificate /etc/asterisk/keys/asterisk.key
Generating RSA private key, 1024 bit long modulus
........++++++
............++++++
e is 65537 (0x010001)
Creating signing request /etc/asterisk/keys/asterisk.csr
Creating certificate /etc/asterisk/keys/asterisk.crt
Signature ok
subject=CN = pbx.example.com, O = My Organization
Getting CA Private Key
Enter pass phrase for /etc/asterisk/keys/ca.key:********
Combining key and crt into /etc/asterisk/keys/asterisk.pem
$ ls -w 1 /etc/asterisk/keys
asterisk.crt
asterisk.csr
asterisk.key
asterisk.pem
ca.cfg
ca.crt
ca.key
tmp.cfg
</pre> </td>
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<p style="margin: 10px 0 0 0"> </p>
<pre style="margin: 10px 0 0 0">We'll use the asterisk.crt, asterisk.key and ca.crt files later.</pre>
<h1 id="ConfiguringAsteriskforWebRTCClients-AsteriskConfiguration" style="margin: 10px 0 0 0; font-size: 24px; font-weight: normal; line-height: 30px; margin: 40px 0 0 0"> <span style="color: rgb(133,120,102);">Asterisk Configuration</span> </h1>
<h2 id="ConfiguringAsteriskforWebRTCClients-ConfigureAsterisk'sbuilt-inHTTPdaemon" style="margin: 10px 0 0 0; font-size: 20px; font-weight: normal; line-height: 30px; margin: 40px 0 0 0; margin-top: 10px">Configure Asterisk's built-in HTTP daemon</h2>
<p style="margin: 10px 0 0 0">To communicate with websocket clients, Asterisk uses its built-in HTTP daemon. Configure <strong>/etc/asterisk/http.conf</strong> as follows:</p>
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<td class="diff-macro-body" style="background-color: #fff;border: 1px solid #dddddd;padding: 10px;; padding: 0px; border-collapse: collapse"> <pre style="margin: 10px 0 0 0; margin-top: 0">[general]
enabled=yes
bindaddr=0.0.0.0
bindport=8088
tlsenable=yes
tlsbindaddr=0.0.0.0:8089
tlscertfile=<your_cert_file>
tlsprivatekey=<your_key_file>
tlscafile=<your_ca_cert_file></pre> </td>
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<p style="margin: 10px 0 0 0">Replace the <code style="font-family: monospace"><your...></code><span> references with the paths to your certificate files. For instance, if you used the procedure above to generate a self-signed cert, </span><code style="font-family: monospace"><your_cert_file></code><span> would be </span><code style="font-family: monospace">/etc/asterisk/keys/asterisk.crt</code><span>, etc.</span> </p>
<p style="margin: 10px 0 0 0">Now start or restart Asterisk and make sure the TLS server is running by issuing the following CLI command:</p>
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<td class="diff-macro-body" style="background-color: #fff;border: 1px solid #dddddd;padding: 10px;; padding: 0px; border-collapse: collapse"> <pre style="margin: 10px 0 0 0; margin-top: 0">*CLI> http show status
HTTP Server Status:
Prefix:
Server: Asterisk/GIT-16-a84c257cd6
Server Enabled and Bound to [::]:8088
HTTPS Server Enabled and Bound to [::]:8089
Enabled URI's:
/test_media_cache/... => HTTP Media Cache Test URI
/guimohdir_rh => HTTP POST mapping
/httpstatus => Asterisk HTTP General Status
/phoneprov/... => Asterisk HTTP Phone Provisioning Tool
/amanager => HTML Manager Event Interface w/Digest authentication
/backups => HTTP POST mapping
/arawman => Raw HTTP Manager Event Interface w/Digest authentication
/manager => HTML Manager Event Interface
/rawman => Raw HTTP Manager Event Interface
/static/... => Asterisk HTTP Static Delivery
/amxml => XML Manager Event Interface w/Digest authentication
/mxml => XML Manager Event Interface
/moh => HTTP POST mapping
/ari/... => Asterisk RESTful API
/ws => Asterisk HTTP WebSocket
<there may be more></pre> </td>
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<p style="margin: 10px 0 0 0">Note that the HTTPS Server is enabled and bound to <code style="font-family: monospace">[::]:8089</code> and that the <code style="font-family: monospace">/ws</code> URI is enabled.</p>
<h2 id="ConfiguringAsteriskforWebRTCClients-ConfigurePJSIP" style="margin: 10px 0 0 0; font-size: 20px; font-weight: normal; line-height: 30px; margin: 40px 0 0 0"> <span style="color: rgb(133,120,102);">Configure PJSIP</span> </h2>
<p style="margin: 10px 0 0 0"> <span style="color: rgb(133,120,102);"> </span>If you're not already familiar with configuring Asterisk's chan_pjsip driver, visit the <a class="confluence-link unresolved" href="#" style="color: #3b73af; text-decoration: none">res_pjsip configuration page</a>.</p>
<h3 id="ConfiguringAsteriskforWebRTCClients-PJSIPWSSTransport" style="margin: 10px 0 0 0; font-size: 16px; line-height: 25px; margin: 30px 0 0 0"> <span style="color: rgb(133,120,102);">PJSIP WSS Transport</span> </h3>
<p style="margin: 10px 0 0 0">Although the HTTP daemon does the heavy lifting for websockets, we still need to define a basic PJSIP Transport for websockets.</p>
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<td style="background-color: #fafafa; padding: 0 0 0 5px; font-size: 12px; text-align: left;; padding: 0px; border-collapse: collapse">/etc/asterisk/pjsip.conf</td>
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<td class="diff-macro-body" style="background-color: #fff;border: 1px solid #dddddd;padding: 10px;; padding: 0px; border-collapse: collapse"> <pre style="margin: 10px 0 0 0; margin-top: 0">[transport-wss]
type=transport
protocol=wss
bind=0.0.0.0
; All other transport parameters are ignored for wss transports.
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<h3 id="ConfiguringAsteriskforWebRTCClients-PJSIPEndpoint,AORandAuth" style="margin: 10px 0 0 0; font-size: 16px; line-height: 25px; margin: 30px 0 0 0"> <span style="color: rgb(133,120,102);">PJSIP Endpoint, AOR and Auth</span> </h3>
<p style="margin: 10px 0 0 0">We now need to create the basic PJSIP objects that represent the client. In this example, we'll call the client <code style="font-family: monospace">webrtc_client</code> but you can use any name you like, such as an extension number. Only the minimum options needed for a working configuration are shown. NOTE: It's normal for multiple objects in pjsip.conf to have the same name as long as the types differ.</p>
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<td class="diff-macro-body" style="background-color: #fff;border: 1px solid #dddddd;padding: 10px;; padding: 0px; border-collapse: collapse"> <pre style="margin: 10px 0 0 0; margin-top: 0">[webrtc_client]
type=aor
max_contacts=1
remove_existing=yes
[webrtc_client]
type=auth
auth_type=userpass
username=webrtc_client
password=webrtc_client ; This is a completely insecure password! Do NOT expose this
; system to the Internet without utilizing a better password.
[webrtc_client]
type=endpoint
aors=webrtc_client
auth=webrtc_client
dtls_cert_file=<your_cert_file>
dtls_private_key=<your_key_file>
dtls_ca_file=<your_ca_cert_file>
webrtc=yes
; Setting webrtc=yes is a shortcut for setting the following options:
; use_avpf=yes
; media_encryption=dtls
; dtls_verify=fingerprint
; dtls_setup=actpass
; ice_support=yes
; media_use_received_transport=yes
; rtcp_mux=yes
context=default
disallow=all
allow=opus
allow=ulaw
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<p style="margin: 10px 0 0 0">An explanation of each of these settings parameters can be found on the <a class="confluence-link unresolved" href="#" style="color: #3b73af; text-decoration: none"><span class="diff-html-changed" id="changed-diff-0" style="background-color: #d6f0ff;">Asterisk </span></a><span class="diff-html-removed" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;"> </span><span class="diff-html-removed" id="removed-diff-0" style="font-size: 100%; background-color: #ffe7e7; text-decoration: line-through;">13 </span><a class="confluence-link unresolved" href="#" style="color: #3b73af; text-decoration: none"><span class="diff-html-added" id="added-diff-0" style="font-size: 100%; background-color: #ddfade;">15 </span><span class="diff-html-changed" id="changed-diff-1" style="background-color: #d6f0ff;">Configuration_res_pjsip</span></a> page. Briefly:</p>
<ul style="margin: 10px 0 0 0">
<li>Declare an endpoint that references our previously-made aor and auth. </li>
<li>Notify Asterisk to expect the AVPF profile (secure RTP)</li>
<li>Setup the DTLS method of media encryption.</li>
<li>Specify which certificate files to use for TLS negotiations with this endpoint and our verification and setup methods.</li>
<li>Enable ICE support</li>
<li>Tell Asterisk to send media across the same transport that we receive it from.</li>
<li>Enable mux-ing of RTP and RTCP events onto the same socket.</li>
<li>Place received calls from this endpoint into an Asterisk <a href="https://wiki.asterisk.org/wiki/display/AST/Dialplan" rel="nofollow" style="color: #3b73af; text-decoration: none">Dialplan</a> context called "default"</li>
<li>And setup codecs by first disabling all and then selectively enabling Opus (presuming that you installed the Opus codec for Asterisk as mentioned at the beginning of this tutorial), then G.711 u-law.</li>
</ul>
<h2 id="ConfiguringAsteriskforWebRTCClients-RestartAsterisk" style="margin: 10px 0 0 0; font-size: 20px; font-weight: normal; line-height: 30px; margin: 40px 0 0 0">Restart Asterisk</h2>
<p style="margin: 10px 0 0 0">Restart Asterisk to pick up the changes and if you have a firewall, don't forget to allow TCP port 8089 through so your client can connect.</p>
<h1 id="ConfiguringAsteriskforWebRTCClients-Wrapup" style="margin: 10px 0 0 0; font-size: 24px; font-weight: normal; line-height: 30px; margin: 40px 0 0 0"> <span style="color: rgb(133,120,102);">Wrapup</span> </h1>
<p style="margin: 10px 0 0 0">At this point, your WebRTC client should be able to register and make calls. If you've used self-signed certificates however, your browser may not allow the connection and because the attempt is not from a normal URI supplied by the user, the user might not even be notified that there's an issue. You <em>may</em> be able to get the browser to accept the certificate by visiting "https://pbx.example.com:8089/ws" directly. This will usually result in a warning from the browser and may give you the opportunity to accept the self-signed certificate and/or create an exception. If you generated your certificate from a pre-existing local Certificate Authority, you can also import that Certificate Authority's certificate into your trusted store but that procedure is beyond the scope of this document.</p>
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